WO2013128696A1 - Digital broadcast method - Google Patents

Digital broadcast method Download PDF

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Publication number
WO2013128696A1
WO2013128696A1 PCT/JP2012/074196 JP2012074196W WO2013128696A1 WO 2013128696 A1 WO2013128696 A1 WO 2013128696A1 JP 2012074196 W JP2012074196 W JP 2012074196W WO 2013128696 A1 WO2013128696 A1 WO 2013128696A1
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mpeg
packet
tts
packets
group
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PCT/JP2012/074196
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French (fr)
Japanese (ja)
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裕美 藤田
永山 英紀
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Nttエレクトロニクス株式会社
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Priority to BR112014021149-3A priority Critical patent/BR112014021149B1/en
Priority to US14/379,246 priority patent/US9357276B2/en
Publication of WO2013128696A1 publication Critical patent/WO2013128696A1/en

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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04NPICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/80Generation or processing of content or additional data by content creator independently of the distribution process; Content per se
    • H04N21/85Assembly of content; Generation of multimedia applications
    • H04N21/854Content authoring
    • H04N21/8547Content authoring involving timestamps for synchronizing content
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60Network streaming of media packets
    • H04L65/61Network streaming of media packets for supporting one-way streaming services, e.g. Internet radio
    • H04L65/611Network streaming of media packets for supporting one-way streaming services, e.g. Internet radio for multicast or broadcast
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60Network streaming of media packets
    • H04L65/65Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60Network streaming of media packets
    • H04L65/70Media network packetisation
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04NPICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/20Servers specifically adapted for the distribution of content, e.g. VOD servers; Operations thereof
    • H04N21/23Processing of content or additional data; Elementary server operations; Server middleware
    • H04N21/238Interfacing the downstream path of the transmission network, e.g. adapting the transmission rate of a video stream to network bandwidth; Processing of multiplex streams
    • H04N21/2389Multiplex stream processing, e.g. multiplex stream encrypting

Definitions

  • the present invention relates to a digital broadcasting method using a communication line.
  • the MPEG-2 TTS data stream is converted into RTP (Real-time Transport Protocol) packets with headers attached to a group of a predetermined number of MPEG-2 TTS packets and transmitted from the transmission side to the communication line.
  • RTP Real-time Transport Protocol
  • null packets were deleted on the transmission side, and non-null MPEG-2 TTS packets were packed up to the upper limit of RTP packets. For this reason, variations occur in the timing of RTP packetization on the transmission side, and the reception side needs to secure a buffer to absorb variations in arrival times of RTP packets. Further, there is a problem that the delay time on the receiving side increases.
  • the present invention has been made in order to solve the above-described problems.
  • the object of the present invention is to eliminate the need for an extra buffer on the receiving side, reduce the delay time on the receiving side, and average the rate of data transfer on the communication line. It is possible to obtain a digital broadcasting method capable of substantially reducing the image quality.
  • the TTS processing unit attaches a time stamp to each packet of the MPEG-2 TS and converts it into MPEG-2 TTS, and the group configuration unit has a predetermined number of the MPEG-2 TTS.
  • a step of collecting packets to form a group a step of a null packet removal unit deleting a null packet from the group, and a transmission unit of the MPEG-2 TTS packet in the group from which the null packet has been deleted. Pre-packed, converted into RTP packets, and transmitted to the communication line.
  • the delay time on the receiving side can be shortened, and the average rate of data transfer on the communication line can be substantially reduced.
  • FIG. 10 is a flowchart of a digital broadcasting method according to Comparative Example 1. It is a figure which shows the example of RTP packetization by the system of the comparative example 2.
  • FIG. 10 is a flowchart of a digital broadcasting method according to Comparative Example 2. It is the figure which compared the example of RTP packetization by each system of Embodiment of this invention and Comparative Examples 1 and 2.
  • FIG. 10 is a flowchart of a digital broadcasting method according to Comparative Example 1. It is the figure which compared the example of RTP packetization by each system of Embodiment of this invention and Comparative Examples 1 and 2.
  • FIG. 1 is a diagram showing a digital broadcasting system according to an embodiment of the present invention. This system is divided into a transmitter 1 and a receiver 2.
  • the receiver 2 receives the RTP packet transmitted from the transmitter 1 via the communication line 3 such as the Internet.
  • the transmitter 1 includes a video / audio encoder (compression / multiplexing device) 4, a TTS processing unit 5, a group configuration unit 6, a Null packet removal unit 7, and a transmission unit 8.
  • FIG. 2 is a diagram showing an example of RTP packetization according to the system of the embodiment of the present invention.
  • FIG. 3 is a flowchart of the digital broadcasting method according to the embodiment of the present invention. The digital broadcasting method of this embodiment will be described with reference to these drawings.
  • the video / audio encoder (compression / multiplexing device) 4 outputs MPEG-2 TS used for normal broadcasting.
  • MPEG-2 TS a Null packet not including a video or the like is inserted in order to adjust the reproduction timing when the original video is encoded by a broadcasting station or the like.
  • the TTS processing unit 5 attaches a time stamp to each MPEG-2 TS packet and converts it to MPEG-2 TTS.
  • the MPEG-2 TS is input to the TTS processing unit 5 (step S1), and the time information at the time of encoding received from the video / audio encoder (compression / multiplexing device) 4 is used as a time stamp for the MPEG-2 TS. (Step S2).
  • the group configuration unit 6 adds an MPEG-2 TTS packet to the group (step S3).
  • the group construction unit 6 repeats steps S1 to S3 until the predetermined number n of MPEG-2 TTS packets is reached (step S4).
  • the group configuration unit 6 passes a group of n packets to the null packet removal unit 7.
  • the transmission unit 8 prepends the MPEG-2 TTS packet in the group from which the Null packet has been deleted, adds a header, converts the packet into an RTP packet, and transmits the packet to the communication line 3 (step S10).
  • FIG. 4 is a diagram illustrating an example of RTP packetization according to the method of the first comparative example.
  • FIG. 5 is a flowchart of the digital broadcasting method according to Comparative Example 1.
  • MPEG-2 TS packets are packed into one RTP packet up to the upper limit without distinguishing Null packets (steps S11 to S14).
  • steps S11 to S14 there is no variation in RTP packetization timing, and in the case of MPEG-2 TS CBR transmission, the reproduction timing can be reproduced on the receiving side in an environment that is not affected by jitter or packet loss.
  • FIG. 6 is a diagram showing an example of RTP packetization by the method of Comparative Example 2.
  • FIG. 7 is a flowchart of the digital broadcasting method according to Comparative Example 2.
  • a time stamp is attached to each packet of MPEG-2 TS, converted to MPEG-2 TTS, and transmitted to communication line 3 (steps S15 to S22).
  • the receiver 2 can rearrange the packets at the correct timing and order. That is, the receiver 2 can accurately reproduce the reproduction timing and order by the time stamp in the MPEG-2 TTS. Therefore, since the transmitter 1 can delete the null packet, the average rate of data transfer on the communication line 3 can be substantially reduced.
  • a predetermined number of packets including Null packets are collected to form a group, and Null packets are deleted from the group and the remaining MPEG-2 TTS packets are left-justified. Convert to RTP packet.
  • RTP packets are generated at regular intervals, no extra buffer is required on the receiving side, and the delay time on the receiving side can be shortened. Even if there is no non-null MPEG-2 TTS packet in one group, the RTP packet is generated only in that group. For this reason, the length of the entire RTP packet is not constant, but there is no problem in the specification.
  • FIG. 8 is a diagram comparing RTP packetization examples according to the respective modes of the embodiment of the present invention and Comparative Examples 1 and 2.
  • Comparative Example 1 the transfer load is large because the NTP packet is included in the RTP packet.
  • Comparative Example 2 the Null packet is deleted to reduce the transfer load, but the RTP packetization timing varies. Therefore, since the arrival interval of the RTP packet is not constant on the receiving side, an extra buffer is required and the delay time increases.
  • the transfer load is at the same level as in Comparative Example 2, and the timing of RTP packetization is at the same level as in Comparative Example 1. Therefore, in this embodiment, no extra buffer is required on the receiving side, the delay time on the receiving side can be shortened, and the average rate of data transfer on the communication line can be substantially reduced.

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  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Signal Processing (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Computer Security & Cryptography (AREA)
  • Two-Way Televisions, Distribution Of Moving Picture Or The Like (AREA)
  • Communication Control (AREA)
  • Television Systems (AREA)
  • Data Exchanges In Wide-Area Networks (AREA)

Abstract

In this digital broadcast method, a TTS conversion processing unit (5) attaches a time stamp to each packet of an MPEG-2 TS (Transport Stream) and converts the MPEG-2 TS to an MPEG-2 TTS (Time-stamped Transport Stream). A group-forming unit (6) gathers a predetermined number of packets of the MPEG-2 TTS to form a group. A null packet removal unit (7) removes null packets from the group. A transmission unit (8) shifts the MPEG-2 TTS packets in the group from which null packets have been removed to the front, converts said MPEG-2 TTS packets into RTP (Real-time Transport Protocol) packets, and transmits the packets to a communication line (3).

Description

デジタル放送方法Digital broadcasting method
 本発明は、通信回線を用いたデジタル放送方法に関する。 The present invention relates to a digital broadcasting method using a communication line.
 インターネット等の通信回線を用いたデジタル放送において、映像と音声を多重化したMPEG-2 TS(Transport Stream)によるハイビジョン放送(高精細動画情報)を行なう場合、要求される転送レートが高い(例えば24Mbps以上)ため、コンテンツに直接関係ない情報(Nullパケット等)を削除して転送レートを下げることが考えられる。 In digital broadcasting using a communication line such as the Internet, when performing high-definition broadcasting (high-definition video information) using MPEG-2 TS (Transport Stream) in which video and audio are multiplexed, a required transfer rate is high (for example, 24 Mbps) For this reason, it is conceivable to delete information (Null packets, etc.) not directly related to the content to lower the transfer rate.
 しかし、MPEG-2 TSの固定ビットレート(CBR: constant bitrate)伝送の場合、受信側で再生タイミングを再現することができるのは、ジッタやパケットロスの影響を全く受けない環境において一定レートで伝送するモデルである。このモデルでNullパケットを削除し平均レートを下げると、伝送情報の受信が困難になる。 However, in the case of MPEG-2 TS constant bit rate (CBR) transmission, playback timing can be reproduced on the receiver side at a constant rate in an environment that is not affected by jitter or packet loss. Model. If null packets are deleted by this model and the average rate is lowered, it becomes difficult to receive transmission information.
 これに対して、MPEG-2 TSの各パケットにタイムスタンプを付けてMPEG-2 TTS(Time-stamped Transport Stream)に変換した後にNullパケットを除去する方法が提案されている(例えば、特許文献1参照)。受信側はMPEG-2 TTS中のタイムスタンプにより再生タイミングを正確に再現できる。従って、送信側はNullパケットを削除することができるため、通信回線でのデータ転送の平均レートを実質的に下げることができる。 On the other hand, there has been proposed a method of removing null packets after attaching a time stamp to each packet of MPEG-2 TS and converting it to MPEG-2 TTS (Time-stamped Transport Transport Stream) (for example, Patent Document 1). reference). The receiving side can accurately reproduce the playback timing using the time stamp in MPEG-2 TTS. Therefore, since the transmitting side can delete the null packet, the average rate of data transfer on the communication line can be substantially reduced.
特開2007-104085号公報JP 2007-104085 A
 MPEG-2 TTSのデータストリームは、所定数のMPEG-2 TTSのパケットを集めたグループにヘッダを付したRTP(Real-time Transport Protocol)パケットに変換されて送信側から通信回線に送信される。従来は、送信側でNullパケットを削除して、NullでないMPEG-2 TTSパケットをRTPパケットの上限一杯まで詰め込んでいた。このため、送信側のRTPパケット化のタイミングにばらつきが生じ、受信側はバッファを確保してRTPパケットの到着時間のばらつきを吸収する必要があった。さらに、受信側の遅延時間が増大するという問題もあった。 The MPEG-2 TTS data stream is converted into RTP (Real-time Transport Protocol) packets with headers attached to a group of a predetermined number of MPEG-2 TTS packets and transmitted from the transmission side to the communication line. In the past, null packets were deleted on the transmission side, and non-null MPEG-2 TTS packets were packed up to the upper limit of RTP packets. For this reason, variations occur in the timing of RTP packetization on the transmission side, and the reception side needs to secure a buffer to absorb variations in arrival times of RTP packets. Further, there is a problem that the delay time on the receiving side increases.
 本発明は、上述のような課題を解決するためになされたもので、その目的は受信側に余分なバッファが不要となり、受信側の遅延時間を短縮でき、通信回線でのデータ転送の平均レートを実質的に下げることができるデジタル放送方法を得るものである。 The present invention has been made in order to solve the above-described problems. The object of the present invention is to eliminate the need for an extra buffer on the receiving side, reduce the delay time on the receiving side, and average the rate of data transfer on the communication line. It is possible to obtain a digital broadcasting method capable of substantially reducing the image quality.
 本発明に係るデジタル放送方法は、TTS化処理部がMPEG-2 TSの各パケットにタイムスタンプを付けてMPEG-2 TTSに変換する工程と、グループ構成部が前記MPEG-2 TTSの所定数のパケットを集めてグループを構成する工程と、Nullパケット除去部が前記グループからNullパケットを削除する工程と、送信部が、前記Nullパケットが削除された前記グループ内の前記MPEG-2 TTSのパケットを前詰めしてRTPパケットに変換して通信回線に送信する工程とを備えることを特徴とする。 In the digital broadcasting method according to the present invention, the TTS processing unit attaches a time stamp to each packet of the MPEG-2 TS and converts it into MPEG-2 TTS, and the group configuration unit has a predetermined number of the MPEG-2 TTS. A step of collecting packets to form a group, a step of a null packet removal unit deleting a null packet from the group, and a transmission unit of the MPEG-2 TTS packet in the group from which the null packet has been deleted. Pre-packed, converted into RTP packets, and transmitted to the communication line.
 本発明により、受信側に余分なバッファが不要となり、受信側の遅延時間を短縮でき、通信回線でのデータ転送の平均レートを実質的に下げることができる。 According to the present invention, no extra buffer is required on the receiving side, the delay time on the receiving side can be shortened, and the average rate of data transfer on the communication line can be substantially reduced.
本発明の実施の形態に係るデジタル放送システムを示す図である。It is a figure which shows the digital broadcasting system which concerns on embodiment of this invention. 本発明の実施の形態の方式によるRTPパケット化の例を示す図である。It is a figure which shows the example of RTP packetization by the system of embodiment of this invention. 本発明の実施の形態に係るデジタル放送方法のフローチャートである。It is a flowchart of the digital broadcasting method which concerns on embodiment of this invention. 比較例1の方式によるRTPパケット化の例を示す図である。It is a figure which shows the example of RTP packetization by the system of the comparative example 1. FIG. 比較例1に係るデジタル放送方法のフローチャートである。10 is a flowchart of a digital broadcasting method according to Comparative Example 1. 比較例2の方式によるRTPパケット化の例を示す図である。It is a figure which shows the example of RTP packetization by the system of the comparative example 2. FIG. 比較例2に係るデジタル放送方法のフローチャートである。10 is a flowchart of a digital broadcasting method according to Comparative Example 2. 本発明の実施の形態及び比較例1,2のそれぞれの方式によるRTPパケット化の例を比較した図である。It is the figure which compared the example of RTP packetization by each system of Embodiment of this invention and Comparative Examples 1 and 2. FIG.
 図1は、本発明の実施の形態に係るデジタル放送システムを示す図である。このシステムは送信機1と受信機2に分かれる。送信機1が送信したRTPパケットをインターネット等の通信回線3を介して受信機2が受信する。送信機1は、映像・音声エンコーダ(圧縮・多重装置)4、TTS化処理部5、グループ構成部6、Nullパケット除去部7、及び送信部8を備える。 FIG. 1 is a diagram showing a digital broadcasting system according to an embodiment of the present invention. This system is divided into a transmitter 1 and a receiver 2. The receiver 2 receives the RTP packet transmitted from the transmitter 1 via the communication line 3 such as the Internet. The transmitter 1 includes a video / audio encoder (compression / multiplexing device) 4, a TTS processing unit 5, a group configuration unit 6, a Null packet removal unit 7, and a transmission unit 8.
 図2は本発明の実施の形態の方式によるRTPパケット化の例を示す図である。図3は本発明の実施の形態に係るデジタル放送方法のフローチャートである。これらの図に沿って本実施の形態のデジタル放送方法を説明する。 FIG. 2 is a diagram showing an example of RTP packetization according to the system of the embodiment of the present invention. FIG. 3 is a flowchart of the digital broadcasting method according to the embodiment of the present invention. The digital broadcasting method of this embodiment will be described with reference to these drawings.
 まず、映像・音声エンコーダ(圧縮・多重装置)4が、通常放送に使われるMPEG-2 TSを出力する。ここで、MPEG-2 TSには、放送局等で元の映像からエンコードされる際に、再生タイミングを調整するために、映像等を含まないNullパケットが挿入されている。 First, the video / audio encoder (compression / multiplexing device) 4 outputs MPEG-2 TS used for normal broadcasting. Here, in the MPEG-2 TS, a Null packet not including a video or the like is inserted in order to adjust the reproduction timing when the original video is encoded by a broadcasting station or the like.
 次に、TTS化処理部5がMPEG-2 TSの各パケットにタイムスタンプを付けてMPEG-2 TTSに変換する。具体的には、TTS化処理部5にMPEG-2 TSが入力され(ステップS1)、映像・音声エンコーダ(圧縮・多重装置)4から受信したエンコード時の時刻情報をタイムスタンプとしてMPEG-2 TSのパケットに付ける(ステップS2)。 Next, the TTS processing unit 5 attaches a time stamp to each MPEG-2 TS packet and converts it to MPEG-2 TTS. Specifically, the MPEG-2 TS is input to the TTS processing unit 5 (step S1), and the time information at the time of encoding received from the video / audio encoder (compression / multiplexing device) 4 is used as a time stamp for the MPEG-2 TS. (Step S2).
 次に、グループ構成部6がMPEG-2 TTSパケットをグループに追加する(ステップS3)。グループ構成部6はMPEG-2 TTSパケットの所定数n個に到達するまでステップS1~S3を繰り返す(ステップS4)。 Next, the group configuration unit 6 adds an MPEG-2 TTS packet to the group (step S3). The group construction unit 6 repeats steps S1 to S3 until the predetermined number n of MPEG-2 TTS packets is reached (step S4).
 次に、グループ構成部6はn個のパケットのグループをNullパケット除去部7に渡す。Nullパケット除去部7はn個のパケットのグループから不要なNullパケットを削除する。具体的には、タイムスタンプが付けられたTSのヘッダを読み(ステップS5)、この中のパケットID(PID)をチェックする(ステップS6)。PIDがNullパケットを意味する“PID=1FFF”である場合は、そのTTSパケットを削除し(ステップS7)、ステップS5に戻る。一方、PIDが“PID=1FFF”でない場合は、そのTTSパケットは有効な情報を含むのでMPEG-2 TTSをRTPペイロード部に追加する(ステップS8)。1つのグループに含まれるMPEG-2 TTSのn個のパケットを処理するまでステップS5~S8を繰り返す(ステップS9)。 Next, the group configuration unit 6 passes a group of n packets to the null packet removal unit 7. The Null packet removal unit 7 deletes unnecessary Null packets from a group of n packets. Specifically, the TS header to which the time stamp is attached is read (step S5), and the packet ID (PID) is checked (step S6). If the PID is “PID = 1FFF” meaning a Null packet, the TTS packet is deleted (step S7), and the process returns to step S5. On the other hand, if the PID is not “PID = 1FFF”, since the TTS packet includes valid information, the MPEG-2 TTS is added to the RTP payload portion (step S8). Steps S5 to S8 are repeated until n packets of MPEG-2 TTS included in one group are processed (step S9).
 次に、送信部8が、Nullパケットが削除されたグループ内のMPEG-2 TTSのパケットを前詰めし、ヘッダを付してRTPパケットに変換して通信回線3に送信する(ステップS10)。 Next, the transmission unit 8 prepends the MPEG-2 TTS packet in the group from which the Null packet has been deleted, adds a header, converts the packet into an RTP packet, and transmits the packet to the communication line 3 (step S10).
 続いて、本実施の形態の効果を比較例1,2と比較して説明する。図4は比較例1の方式によるRTPパケット化の例を示す図である。図5は比較例1に係るデジタル放送方法のフローチャートである。比較例1では、Nullパケットを区別せずにMPEG-2 TSパケットを1つのRTPパケットに上限一杯まで詰め込む(ステップS11~S14)。これによりRTPパケット化のタイミングにばらつきが生じないため、MPEG-2 TSのCBR伝送の場合、ジッタやパケットロスの影響を全く受けない環境では、受信側で再生タイミングを再現することができる。 Subsequently, the effect of the present embodiment will be described in comparison with Comparative Examples 1 and 2. FIG. 4 is a diagram illustrating an example of RTP packetization according to the method of the first comparative example. FIG. 5 is a flowchart of the digital broadcasting method according to Comparative Example 1. In the first comparative example, MPEG-2 TS packets are packed into one RTP packet up to the upper limit without distinguishing Null packets (steps S11 to S14). As a result, there is no variation in RTP packetization timing, and in the case of MPEG-2 TS CBR transmission, the reproduction timing can be reproduced on the receiving side in an environment that is not affected by jitter or packet loss.
 一方、インターネット等の通信回線を用いたデジタル放送において、ハイビジョン放送を行なうことを考える。ハイビジョン放送では要求される転送レートが高く、これを下げるためNullパケットを削除することが考えられるが、受信側で再生タイミングの再現が困難になる。よって、Nullパケットを削除することができないため、比較例1はデジタル放送への適用が難しい。 On the other hand, consider high-definition broadcasting in digital broadcasting using communication lines such as the Internet. In high-definition broadcasting, the required transfer rate is high, and it is conceivable to delete a null packet in order to lower this, but it is difficult to reproduce the reproduction timing on the receiving side. Therefore, since the Null packet cannot be deleted, it is difficult to apply Comparative Example 1 to digital broadcasting.
 図6は比較例2の方式によるRTPパケット化の例を示す図である。図7は比較例2に係るデジタル放送方法のフローチャートである。比較例2では、MPEG-2 TSの各パケットにタイムスタンプを付けてMPEG-2 TTSに変換して通信回線3に送信する(ステップS15~S22)。このため、受信機2で受け取ったMPEG-2 TTSのパケットのタイミングや順序が送信側からずれても、受信機2は正しいタイミングや順序でパケットを再配列することができる。即ち、受信機2はMPEG-2 TTS中のタイムスタンプにより再生タイミングや順序を正確に再現できる。従って、送信機1はNullパケットを削除することができるため、通信回線3でのデータ転送の平均レートを実質的に下げることができる。 FIG. 6 is a diagram showing an example of RTP packetization by the method of Comparative Example 2. FIG. 7 is a flowchart of the digital broadcasting method according to Comparative Example 2. In Comparative Example 2, a time stamp is attached to each packet of MPEG-2 TS, converted to MPEG-2 TTS, and transmitted to communication line 3 (steps S15 to S22). For this reason, even if the timing and order of MPEG-2 TTS packets received by the receiver 2 deviate from the transmission side, the receiver 2 can rearrange the packets at the correct timing and order. That is, the receiver 2 can accurately reproduce the reproduction timing and order by the time stamp in the MPEG-2 TTS. Therefore, since the transmitter 1 can delete the null packet, the average rate of data transfer on the communication line 3 can be substantially reduced.
 しかし、比較例2では、Nullパケットを削除して、NullでないMPEG-2 TTSパケットをRTPパケットの上限一杯まで詰め込む。このため、送信側のRTPパケット化のタイミングにばらつきが生じ、受信側はバッファを確保してRTPパケットの到着時間のばらつきを吸収する必要がある。さらに、受信側の遅延時間が増大する。 However, in Comparative Example 2, the Null packet is deleted, and the non-Null MPEG-2 TTS packet is packed up to the upper limit of the RTP packet. For this reason, variations occur in the timing of RTP packetization on the transmission side, and the reception side needs to secure a buffer to absorb variations in arrival times of RTP packets. Furthermore, the delay time on the receiving side increases.
 これに対して、本実施の形態では、Nullパケットも含めて所定数のパケットを集めてグループを構成し、そのグループからNullパケットを削除して残ったMPEG-2 TTSのパケットを前詰めしてRTPパケットに変換する。これにより、RTPパケットが一定間隔で生成されるため、受信側に余分なバッファが不要となり、受信側の遅延時間を短縮できる。なお、1つのグループ中にNullでないMPEG-2 TTSパケットが一つも存在しない場合でも、そのグループだけでRTPパケット化する。なお、このためにRTPパケット全体の長さは一定ではなくなるが、仕様上は問題ない。 On the other hand, in this embodiment, a predetermined number of packets including Null packets are collected to form a group, and Null packets are deleted from the group and the remaining MPEG-2 TTS packets are left-justified. Convert to RTP packet. Thereby, since RTP packets are generated at regular intervals, no extra buffer is required on the receiving side, and the delay time on the receiving side can be shortened. Even if there is no non-null MPEG-2 TTS packet in one group, the RTP packet is generated only in that group. For this reason, the length of the entire RTP packet is not constant, but there is no problem in the specification.
 図8は本発明の実施の形態及び比較例1,2のそれぞれの方式によるRTPパケット化の例を比較した図である。比較例1では、Nullパケットも含めてRTPパケット化しているため、転送負荷が大きい。比較例2では、Nullパケットを削除して転送負荷を軽減しているが、RTPパケット化のタイミングにばらつきが生じる。よって、受信側でRTPパケットの到着間隔も一定でなくなるため、余分なバッファが必要となり、遅延時間が増大する。これに対して、本実施の形態は、転送負荷は比較例2と同レベルであり、RTPパケット化のタイミングは比較例1と同レベルである。よって、本実施の形態では、受信側に余分なバッファが不要となり、受信側の遅延時間を短縮でき、通信回線でのデータ転送の平均レートを実質的に下げることができる。 FIG. 8 is a diagram comparing RTP packetization examples according to the respective modes of the embodiment of the present invention and Comparative Examples 1 and 2. In Comparative Example 1, the transfer load is large because the NTP packet is included in the RTP packet. In Comparative Example 2, the Null packet is deleted to reduce the transfer load, but the RTP packetization timing varies. Therefore, since the arrival interval of the RTP packet is not constant on the receiving side, an extra buffer is required and the delay time increases. On the other hand, in the present embodiment, the transfer load is at the same level as in Comparative Example 2, and the timing of RTP packetization is at the same level as in Comparative Example 1. Therefore, in this embodiment, no extra buffer is required on the receiving side, the delay time on the receiving side can be shortened, and the average rate of data transfer on the communication line can be substantially reduced.
1 送信機
2 受信機
3 通信回線
5 TTS化処理部
6 グループ構成部
7 Nullパケット除去部
8 送信部
DESCRIPTION OF SYMBOLS 1 Transmitter 2 Receiver 3 Communication line 5 TTS process part 6 Group structure part 7 Null packet removal part 8 Transmitter part

Claims (1)

  1.  TTS化処理部がMPEG-2 TS(Transport Stream)の各パケットにタイムスタンプを付けてMPEG-2 TTS(Time-stamped Transport Stream)に変換する工程と、
     グループ構成部が前記MPEG-2 TTSの所定数のパケットを集めてグループを構成する工程と、
     Nullパケット除去部が前記グループからNullパケットを削除する工程と、
     送信部が、前記Nullパケットが削除された前記グループ内の前記MPEG-2 TTSのパケットを前詰めしてRTP(Real-time Transport Protocol)パケットに変換して通信回線に送信する工程とを備えることを特徴とするデジタル放送方法。
    A step in which a TTS processing unit attaches a time stamp to each packet of MPEG-2 TS (Transport Stream) and converts the packet to MPEG-2 TTS (Time-stamped Transport Stream);
    A group forming unit collecting a predetermined number of packets of the MPEG-2 TTS to form a group;
    A step of a null packet removing unit deleting a null packet from the group;
    A transmitting unit comprising a step of prepacking the MPEG-2 TTS packet in the group from which the Null packet has been deleted, converting the packet into an RTP (Real-time Transport Protocol) packet, and transmitting the packet to a communication line. A digital broadcasting method characterized by the above.
PCT/JP2012/074196 2012-03-01 2012-09-21 Digital broadcast method WO2013128696A1 (en)

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